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Course Outline
Part I: Introduction
- Introduction
- Historical context and objectives
- Categories of Voice over Internet Protocol (VoIP) and technological progression
- Core principles of the Session Initiation Protocol (SIP)
- SIP standardization framework (RFC 3261 and associated standards)
- System Architecture
- User Agent (UA) components
- Essential server types: Registrar, Location, Proxy, and Redirect
- Application server functions
- Entity identification and addressing schemes
- SIP protocol topology
- Server Operations
- User registration processes
- Proxy and Redirect server functionalities
- Stateless versus stateful proxy mechanisms
- Location services
- SRV records and Domain Name System (DNS) integration
- URI, URL, URN formats; ENUM services; and NAPTR records
- SIP Signalling Mechanisms (including Instant Messaging & Presence – IMP extensions)
- Message architecture
- Request types
- Response codes
- Illustrative call procedure
- Headers and parameters
- IMP operational models
- Session Description Protocol (SDP)
- Media description formats
- Standardized codec inventory
- Session negotiation protocols
- Signalling Flows – SIP
- SIP session establishment – primary RFC 3261 example
- Representative call scenarios
- Conferencing and IP Private Branch Exchange (PBX) integration
- Dynamic media modification during sessions
- Utilization of IMP services
- Routing of SIP Requests and Responses
- VIA header processing
- ROUTE and RECORD-ROUTE headers
- SIP to Public Switched Telephone Network (PSTN) Interworking
- SIP-T and SIP-I standards
- Early media and SIP trunking capabilities
- SIP-PSTN signalling integration
- Security Considerations in SIP
- Securing SIP, RTP, and RTCP traffic
- Common implementations for Secure SIP
- Operational Challenges and Future Outlook
- NAT and firewall traversal techniques
- Quality of Service (QoS) implementation
- SIP and SDP integration within 3GPP IMS architecture
- Summary and Discussion
Part II: Practical Application
- Deploying SIP in Local Area Network Environments: XLite SIP UA + Asterisk
- Establishing Asterisk accounts with basic dial plan configuration
- Configuring XLite SIP User Agent (DTMF, codecs, NAT, RTP, timer, registration) and SIP endpoints (Polycom, Gigaset, Yealink, Linphone)
- Managing registration, call initiation, and reception
- Peer-to-peer calling with Linphone
- Analyzing SIP Signalling via Wireshark
- Server configuration protocols
- Capture of SIP signalling and RTP media streams
- SIP packet analysis and isolation of specific call records
- Troubleshooting voice quality issues: Jitter buffer management, DTMF retrieval (RFC 2833, INFO), codec/DNMF diagnostics (transcoding, GSM failures, tone duplication)
- VoIP monitoring tools
- SDP, Instant Messaging, and Presence (IM&P)
- SDP parameters and attributes
- SUBSCRIBE, PUBLISH, and MESSAGE SIP methods
- Practical IM&P exercises using XLite and Linphone
- Examining SIP Call Flows
- SIP Registration via DNS
- SIP SRV record mechanics
- Endpoint registration utilizing DNS-SRV
- Call Flow Analysis with DNS
- Analyzing SIP signalling traces in Wireshark
- Troubleshooting DNS timeouts and latency issues
- SIP Registration via DNS
- SIP Trunking
- Establishment of test SIP trunks
- Troubleshooting for Denial of Service (DoS), Distributed Denial of Service (DDoS), fraud, and calls per second (CPS) limits
- SIP Security Measures
- Implementing SIP security with IPsec
- Securing SIP communications
- Risks associated with IP telephony fraud
- Strategies for preventing DDoS and other attack vectors
- Deploying SIP-Based VoIP Services
- Switch configuration procedures
- SIP client setup and registration protocols
- Software Infrastructure
- Asterisk PBX / FreeSWITCH softswitch / Cisco Call Manager
- Linux CentOS environment
- TDM-to-IP drivers
- Softphone applications (XLite, Linphone)
- Hardware Components
- Server hardware
- TDM-to-IP cards/gateways
- Hardphones (Polycom, Gigaset, Yealink)
- Softphone and Hardphone Management
- Configuration Settings
- Codec selection
- User credentials, SIP server/proxy details, and port assignments
- Operational Signalling for:
- Three-way calling
- Call forwarding mechanisms
- Attendant call transfers
- Message Waiting Indication (MWI) and Busy Lamp Field (BLF)
- Yealink automated provisioning
- Vendor-specific limitations
- Configuration Settings
- SIP and Network Address Translation (NAT) Challenges
- NAT types and structural components
- STUN (Simple Traversal of UDP Through NATs) protocol
- Troubleshooting VoIP Call Quality
- Diagnostics for connected calls with missing media
- Critical QoS parameters
- Latency, jitter, and playback buffer size
- VoIP quality metrics
- RTCP analysis for delay and jitter
- Mean Opinion Score (MOS) per ITU-T G.107 E-model
- Quality monitoring tools (e.g., Voipmonitor)
- Cloud-Based IP Telephony Solutions
- Summary and resolution of participant-submitted SIP and VoIP issues
21 Hours
Testimonials (7)
Good contact with the trainer
Radziszewski - EduBroker Sp. z o.o.
Course - SIP protocol in VoIP
Detailed analysis of the provided traces
Krzysztof - EduBroker Sp. z o.o.
Course - SIP protocol in VoIP
examples presented
Slawomir - NBP
Course - SIP protocol in VoIP
The instructor patient to explain and show the step on each lab session.
CHEE WAI KAN - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP
Trainer can explain and give example for questions asked
SHAMSANI ABU BAKAR - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP
Very informative and detail explanation
Syahrifendi Lukmanilhakim - Maxis Broadband Sdn Bhd
Course - SIP protocol in VoIP
The tools and the contents of the slide.